In packet voice systems, as in any packet-based system, periods of traffic congestion may result. This traffic congestion adds delay to the packet system and, as such, impacts the ability of people to effectively carry on a conversation. Congestion also results in packet losses due to buffer overflow, and creates degradation in quality of voice communication.
One approach to alleviating traffic congestion is to selectively drop entire voice packets. Generally, the selectively dropped voice packets represent "less-important" speech such as "semi-silence" (e.g., see Nanying Yin, San-Qi Li, and Thomas E. Stern, "Congestion Control for Packet Voice by Selective Packet Discarding," IEEE Trans. on Communications, May 1990, Vol. 38, No. 5; Kotikalapudi Sriram, R. Scott McKinney, and Mostafa Hashem Sherif, "Voice Packetization and Compression in Broadband ATM Networks," IEEE Journal on Selected Areas in Communications, Vol. 9, No. 3, April 1991; and David W. Petr, Luiz A DaSilva, Jr., Victor S. Frost, "Priority Discarding of Speech in Integrated Packet Networks," IEEE Journal on Selected Areas in Communications, Vol. 7, No. 5, June 1989). (For completeness only, it should be noted that there are also other alternatives such as ITU (International Telecommunications Union) Embedded Adaptive Differential Pulse Code Modulation (ADPCM) Standard G.727, which describes a method for dropping the least significant bits of a voice sample within a packet.)
When entire voice packets are dropped, subjective voice quality is affected. For example, it is well known that subjective voice quality degradation is virtually imperceptible for a random packet loss rate of up to 1 in 100 for wavefrom encoded speech such as pulse code modulation (CM) and ADPCM (e.g., see J. G. Gruber and N. Le, "Performance requirements for integrated voice/data networks," IEEE Journal on Selected Areas in Communications, December 1983, pp. 981-1005; and N. S. Jayant and S. W. Christensen, "Effects of packet losses in waveform coded speech and improvements due to an odd-even sample interpolation," IEEE Trans. on Communications, February 1981, pp. 101-109).
Unfortunately, selective discarding of entire voice packets as a function of the type of speech, e.g., semi-silence, may cause consecutive packets from the same voice source to be discarded. This phenomenon is shown in FIG. 1, which illustrates the nature of the packet voice arrival process (from multiple sources). During periods of congestion, the packets from a few sources tend to arrive at periodic intervals at the tail ends of bursts of packets from all sources. This is shown in FIG. 1 by burst 1, burst 2, and burst 3, and circles 11, 12, and 13. Consequently, in these periods these few sources experience excessive loss of consecutive packets due to buffer overflow, while packets of other sources do not experience such excessive loss. The loss of consecutive packets further degrades subjective voice quality and may become noticeable to the receiving party. As such, to avoid the perceptive quality degradation due to consecutive packet loss, the packet system is often designed for a much lower packet loss rate (e.g., about 1 in 1000 (or less) in the case of ADPCM encoded speech) for the voice quality degradation due to consecutive packet losses to remain imperceptible to the listener. Such a low packet loss rate requirement, of course, results in operating the system at much lower engineered traffic load and hence higher cost of the packet system.